09-26-2012 11:56 PM - edited 03-16-2019 01:24 PM
Hi,
Currently i have an scenario with
alcatel (h323) -> 2921 -> SIP Trunk -> Carrier Softswitch
When i call to a PSTN number form an alcatel extension by going through the SIP Trunk from the alcatel extension i herd ringback, and on the PSTN phone rings, when the phone is answered then there is a silence on the call and then sudently drop the call.
I collected the sip debg info
2012-09-14 10:08:43 Local7.Notice 172.16.4.2 5825010: *Sep 14 09:49:34: %SYS-5-CONFIG_I: Configured from console by sistlajo on vty0 (172.16.1.43)
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825011: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825012: The Call Setup Information is:
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825013: Call Control Block (CCB) : 0x9AA6D58
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825014: State of The Call : STATE_DEAD
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825015: TCP Sockets Used : NO
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825016: Calling Number : 3324004900
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825017: Called Number : 37700028
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825018: Source IP Address (Sig ): 10.15.0.22
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825019: Destn SIP Req Addr:Port : 10.255.252.134:5060
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825020: Destn SIP Resp Addr:Port : 10.255.252.134:5060
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825021: Destination Name : 10.255.252.134
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825022:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825023: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPIMediaCallInfo:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825024: Number of Media Streams: 1
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825025: Media Stream : 1
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825026: Negotiated Codec : g711alaw
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825027: Negotiated Codec Bytes : 160
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825028: Nego. Codec payload : 8 (tx), 8 (rx)
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825029: Negotiated Dtmf-relay : 6
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825030: Dtmf-relay Payload : 101 (tx), 101 (rx)
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825031: Source IP Address (Media): 10.15.0.22
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825032: Source IP Port (Media): 0
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825033: Destn IP Address (Media): 172.16.250.29
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825034: Destn IP Port (Media): 31100
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825035: Orig Destn IP Address:Port (Media): [ - ]:0
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825036:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825037: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825038: Disconnect Cause (CC) : 16
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825039: Disconnect Cause (SIP) : 481
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825040:
Here is the meaningful part of my configuration:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711ulaw
h323
call preserve
modem passthrough nse codec g711ulaw
--More-- sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
voice class codec 10
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
voice class codec 11
codec preference 1 g729r8
codec preference 2 g711alaw
codec preference 3 g711ulaw
codec preference 4 g729br8
!
voice class codec 12
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
!
voice class h323 1
--More-- h225 timeout tcp establish 3
!
!
!
voice translation-rule 10
rule 1 /\([^9].*\)/ /9\1/
!
voice translation-rule 20
rule 1 /^9/ //
!
voice translation-rule 21
rule 1 /..../ /3324004900/
!
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 172.16.4.2 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.4.2
!
interface GigabitEthernet0/1
ip address 10.15.0.22 255.255.255.0
duplex auto
--More-- speed auto
media-type rj45
no cdp enable
!
ip route 0.0.0.0 0.0.0.0 172.16.4.1
ip route 10.255.252.134 255.255.255.255 10.15.0.254
ip route 172.16.250.0 255.255.255.0 10.15.0.254
ip route 172.16.253.0 255.255.255.0 10.15.0.254
!
dial-peer voice 4011 voip
description ~-~-~-~-~-~-~-Dir 937700028~-~-~-~-~-~-~-
translation-profile outgoing SalidaSIP2
preference 2
destination-pattern 937700028
session protocol sipv2
session target ipv4:10.255.252.134
session transport udp
voice-class codec 12
dtmf-relay rtp-nte h245-alphanumeric h245-signal cisco-rtp sip-kpml sip-notify
no vad
as you can notice i hardcoded the phone number i'm trying to reach,
Hope to find help soon, on advice thanks!
Regards,
Solved! Go to Solution.
09-27-2012 12:29 PM
Hi Claudio
Just some thoughts
its better to have (incoming called number .) for calls coming from alcatel through h323 (h323 dial peer)and also the same for calls coming from sip trunk(sip dial peer)
1)Are you able to ping the isp ip
2) incoming calls are working?
3) so the provider is using a calling number athentication(or no authentication at all)
4) The calling number is what isp expecting?
5) did you ask isp to be involed on that?Help with trace etc?
Could you pls answer the above
We will be waiting for the debug ccsip messages
Regards
chrysostomos
10-02-2012 01:08 PM
Is this the called number 37980899 ?
Can you explain the set up better...Is this diagram correct?
Alcatel---h323--->CUBE--sip trunk-->-ITSP
Where are you making the call from..Is it from alcatel to Carried? or carrier to alcatel?
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
09-27-2012 12:19 AM
Hi
I am wondering if the sip trunk is register with the isp
show sip-ua registration status
debug ccsip messages
One question
calls from alcatel are g711 or g729
what protocol have the alcatel phones(sip?)
09-27-2012 06:45 AM
hi, calls from alcatel are g711alaw and the alcatel phones are analog connected to the pbx, which send the calls to the voice gateway through h323 protocol.
No sip-ua registration because everything is on dial-peers configuration, carrirer is not having authentication.
I noticed there is also a SIP Cause code 481 also, but i think this could be caused depending if the people on the pstn phone answers when the phone rings, if this is distracting lets focus on 487.
2012-09-14 09:57:52 Local7.Notice 172.16.4.2 5825009: *Sep 14 09:38:42: %SYS-5-CONFIG_I: Configured from console by sistlajo on vty0 (172.16.1.43)
2012-09-14 10:08:43 Local7.Notice 172.16.4.2 5825010: *Sep 14 09:49:34: %SYS-5-CONFIG_I: Configured from console by sistlajo on vty0 (172.16.1.43)
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825011: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825012: The Call Setup Information is:
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825013: Call Control Block (CCB) : 0x9AA6D58
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825014: State of The Call : STATE_DEAD
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825015: TCP Sockets Used : NO
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825016: Calling Number : 3324004900
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825017: Called Number : 37700028
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825018: Source IP Address (Sig ): 10.15.0.22
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825019: Destn SIP Req Addr:Port : 10.255.252.134:5060
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825020: Destn SIP Resp Addr:Port : 10.255.252.134:5060
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825021: Destination Name : 10.255.252.134
2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825022:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825023: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPIMediaCallInfo:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825024: Number of Media Streams: 1
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825025: Media Stream : 1
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825026: Negotiated Codec : g711alaw
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825027: Negotiated Codec Bytes : 160
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825028: Nego. Codec payload : 8 (tx), 8 (rx)
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825029: Negotiated Dtmf-relay : 6
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825030: Dtmf-relay Payload : 101 (tx), 101 (rx)
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825031: Source IP Address (Media): 10.15.0.22
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825032: Source IP Port (Media): 0
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825033: Destn IP Address (Media): 172.16.250.29
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825034: Destn IP Port (Media): 31100
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825035: Orig Destn IP Address:Port (Media): [ - ]:0
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825036:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825037: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825038: Disconnect Cause (CC) : 16
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825039: Disconnect Cause (SIP) : 481
2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825040:
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825041: *Sep 14 10:22:41: //2141792/80EFAF09D636/SIP/Call/sipSPICallInfo:
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825042: The Call Setup Information is:
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825043: Call Control Block (CCB) : 0x9AA6D58
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825044: State of The Call : STATE_DEAD
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825045: TCP Sockets Used : NO
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825046: Calling Number : 3324004900
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825047: Called Number : 37700028
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825048: Source IP Address (Sig ): 10.15.0.22
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825049: Destn SIP Req Addr:Port : 10.255.252.134:5060
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825050: Destn SIP Resp Addr:Port : 10.255.252.134:5060
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825051: Destination Name : 10.255.252.134
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825052:
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825053: *Sep 14 10:22:41: //2141792/80EFAF09D636/SIP/Call/sipSPICallInfo:
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825054: Disconnect Cause (CC) : 16
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825055: Disconnect Cause (SIP) : 487
2012-09-14 10:41:51 Local7.Debug 172.16.4.2 5825056:
Thank you for your quick response and regards,
Claudio.
09-27-2012 07:01 AM
Hi
Two questions
What is the ip:172.16.250.29
What is the ip :10.255.252.134
Pls send debug ccsip messages, debug ccsip errors in VG
what is the version of your ios?
Regards
09-27-2012 11:53 AM
Thank you Chrys,
Version 15.0(1)M4
The ip address 172.16.250.29 - The RTP is passed by an rtp forwarder on the carrier side. Seems they handle the RTP on another equipment. I don't think this is an impact, i have a CUCM also using this gateway and sip trunk but on this scenario doesn't have problems.
Currently to place the test again i need to modify the scenario from the Alcatel side and this impacts the user, I'll try to get the debugs as soon as posible.
Regards,
Claudio.
09-27-2012 12:29 PM
Hi Claudio
Just some thoughts
its better to have (incoming called number .) for calls coming from alcatel through h323 (h323 dial peer)and also the same for calls coming from sip trunk(sip dial peer)
1)Are you able to ping the isp ip
2) incoming calls are working?
3) so the provider is using a calling number athentication(or no authentication at all)
4) The calling number is what isp expecting?
5) did you ask isp to be involed on that?Help with trace etc?
Could you pls answer the above
We will be waiting for the debug ccsip messages
Regards
chrysostomos
09-27-2012 12:53 PM
Thank you Chrys,
1.- im able to ping the isp
2.- incomming calls are on another trunk, this is just for outbound
3.- no authentication on the carrier side for sip
4.- yes, they requested us to change the 4 digit extension format (ej. 3811@ip address for a 10 digit number (ej 3312000122@ipaddress)
5.- the carrier is involved
I'm preparing the scenario to collect the debugs.
Regards,
09-27-2012 02:44 PM
Hi,
I taked the debugs from the gateway, thank you for your patience, the callflow is:
alcatel->voice gateway->sip trunk
The caller keeps listening ringback even if the call.
The pstn phone listen silence for a while when the call is answered, but suddently listen fast busy as if the call where disconnected.
Best regards,
Claudio.
09-28-2012 01:01 PM
Hi Chris, i've noticed about a response from you this morning about changing a parameter on the Alcatel, but now that i'm on site i can't get your post to retrieve the value that you suggested, can you repeat me the value that you suggested on your last/deleted post?
Regards,
09-28-2012 11:23 PM
Hi Claudio
Config Alcatel for Fast Start on the H323 connection.
Sent from Cisco Technical Support iPhone App
10-01-2012 10:21 AM
Hi,
Configured alcatel for Destination: Gateway Fast Start on the Alcatel OmniPCX
Also tryied configured on gateway the following lines:
voice service voip
h323
call start fast
but the result is the same.
On the destination number there is just silence.
Regards,
Claudio.
10-01-2012 10:59 AM
Hi claudio
Did you check without the fast start enable in the cube?
Can you send another one debug?
10-01-2012 01:26 PM
10-01-2012 02:04 PM
Hi
First test:
ip route 10.15.0.22 255.255.255.255 10.15.0.254
Test one call with debug
Second test:
Remove the bind commans into sip
Test one call with debug
10-01-2012 02:08 PM
Hi Claudio,
I have looked at your traces and here is what I see..
1. Your CUBE sends an invite with delayed offer, that means you do not send any SDP in your invite
2. In this scenario, you cant enable fast start on your h323 trunk becasue cube does not support delayed offer to fast start support.
3. Something is not quite wright with your sip integration to alcatel..
Alcaltel does not send the required sip messages such as "trying", 183 ringing..Insteand alcatel sends 2oo ok without any SDP parameter...then sends 183 with media change, and sends it multiple times because cube does not send any ACK to it. I belive the reason is because cube does not understand what the message is for..
So here is my advice
1. Try and enable early offer on your cube
voice service voip
sip
early-offer forced.
Then test again..If this doesnt work, you will need to check CCO for how to integrate your version of Alcatel with CUCM and CUBE
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
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